The invention relates generally to audio sampling rate conversion FIR filters and more particularly to low pass audio sampling rate conversion FIR filters that perform interpolation and/or decimation by a factor of two such as 1:2 interpolation and/or 2:1 decimation.
The trend in multimedia computer systems is to have numerous audio sources, such as CD's, DVD's, stored audio files and digital satellite broadcast signals that can be mixed and need to be output through the computer to speakers. The audio signal to the speakers is typically specified at a different sampling rate than the audio sampling rate from the sources. In digital audio applications, standard sampling rates and conversions may be as shown below in Table 1.
TABLE 1 ______________________________________ original sampling rate (kHz) new sampling rate (kHz) ______________________________________ 11.025 22.05 22.05 44.1 8 16 16 32 ______________________________________
The individual data stream must each be converted to a common sampling rate of typically 48 kwps (kilowords per second). Mixing of these streams from the various sources may generally be accomplished by taking a stream at 8 kwps and converting it by a factor of two to 16 kwps and mixing the data with the stream at 16 kwps. In the same way, a 16 kwps stream is converted by a factor of two to 32 kwps stream and can be mixed with a stream at 32 kwps. The rate may also be in bytes per second or other suitable data size.
For example, where audio from a source or channel has been sampled at 8 kwps and the signal must be mixed with audio from another source that outputs sampled signals at a rate twice as much, such as 16 kwps, the audio signal must undergo 1:2 interpolation to provide twice as many samples for the same signal. Conversely, where the source audio has been sampled at twice the rate than audio it is to be mixed with, the signal must undergo filtering and be decimated to provide one-half as many samples for the same signal.
With many audio sources and a limited amount of computer processing capability and memory in multimedia systems (and audio telecommunication systems), efficient sampling rate conversion is necessary. Generally, during interpolation or decimation, a digital filter must perform multiplications for each filter coefficient to determine a corresponding signal value. Each multiplication step requires processing time. A typical sampling rate converter uses a half-band FIR filter for performing all of the interpolation and decimation. Such filters, as known in the art, use nearly one-half the number of filter coefficients compared to a normal FIR filter while providing a similar signal to noise quality level. For example, where a 96 dB S/N ratio is desired, a normal FIR filter may require the use of 60 filter coefficients whereas the halfband FIR filter may only require about 30 coefficients. However, such filters still require input samples with zero values inserted so the number of multiplication operations is still higher than desired. Operation of a standard half band filter is shown in FIG. 1.
Another type of rate converter uses a poly-phase FIR filter (or filters) to perform all interpolation or decimation. The poly-phase FIR filter is more efficient than a conventional FIR filter because it skips the zero values inserted. However, such filters still require a large number of coefficients so that additional audio conversion will be slower than necessary to accommodate mixing of numerous audio sources.
Consequently, there exists a need for an improved audio sampling rate conversion filter that performs interpolation and/or decimation by a factor of two such as 1:2 interpolation and/or 2:1 decimation. It would be advantageous if such a filter required fewer coefficients and processing time to convert audio signals.